In previous years, Voice over internet protocol (VoIP) has become an important application and is expected to carry more and more voice traffic over TCP/IP networks.
In such Internet protocol (IP)-based voice communications systems, typically a voice waveform of a user is sliced in time, compressed by a voice coder, packetized, and transmitted to other users. Due to the inherent nature of IP networks and real-time constraint of human voice communications, it is common to lose voice packets during transmission or that late voice packets are discarded even if they are received, resulting in degraded voice quality. Mobile and WIFI networks usually make the situation worse in many cases. Thus, accurate real-time monitoring of voice quality is an essential feature for analysis, management, and optimization of voice communication systems.
A typical voice quality monitoring system adopts a scheme that analyzes packet loss information, such as packet loss rate and loss patterns (e.g., if the losses are random or of a bursty nature), as it provides a simple and computationally inexpensive way to estimate voice quality. This scheme is known as a modified E-model. However, these systems suffer from low accuracy in estimating voice quality since they do not take the content (e.g. the payload) of the lost voice packets into consideration when estimating the voice quality.
More accurate voice quality estimation may be achieved by analyzing voice waveforms after fully decoded all the packets and other data sent in the VoIP call (e.g. ITU-T P.563, ANSI ANIQUE+). However, this approach requires extensive computation for analyzing the voice waveforms. Moreover, this approach discards important packet loss statistic information available at packet level.
Thus, it is desirable to have a voice quality monitoring system utilizing both packet loss information and speech waveform information without an expensive full decoding process.
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